Techniques for transmission of real-time signals such as streaming audio or video, voice over IP (VoIP) or videoconferencing media streams in Internet Protocol (IP) based networks are in high demand due to prospect of cost savings derived from the use of datacommunications equipment based on packet switched transmission instead of circuit-switched transmission networks.
To accomplish this task, certain Quality of Service (QoS) problems need to be solved. One such problem is the variability of transmission delays in IP networks. Such a variability leads to variation of packet interarrival times, even if the packets had been transmitted at equal intervals. Such a variation in interarrival times leads to burstiness of data stream bandwidth. Stated in other words, although data packets are transmitted by a data source (e.g. a communication terminal or network element) in regular/constant intervals, these intervals might be subject to variation due to the transmission path for/processing of the data packets within the network. As a result, the data packets are received at the destination (e.g. a communication terminal or network element) in clusters or groups in which an interval between packets is shorter than the interval between the sent packets, and/or wherein an interval between consecutively received data packets is longer than the interval between the sent packets. This phenomenon is also referred to as burstiness.
This kind of variability can be reduced with traffic shaping. Traffic shaping is accomplished by using a traffic shaper device with a configurable output bandwidth. Note that throughout the specification, a bandwidth (input or output bandwidth) of a device related to data packet transmission is intended to mean a corresponding data rate of the sent/received data packets (i.e. number of data and/or data packets per unit time).
If such a high number of packets arrives at the input of a traffic shaper device that they exceed the configured output bandwidth, they may be enqueued in a buffer inside the traffic shaper device and transmitted later so that an output bandwidth limit criterion is fulfilled. Due to reduction of burstiness of streams, traffic shaping at network edges makes aggregate bandwidth easier to manage. A reduction of burstiness is accomplished since the data packet stream output from the traffic shaper device is output at regular intervals (i.e. with the output bandwidth of the traffic shaper device) so that formation of data packet clusters is suppressed or prevented.
Queuing, i.e. buffering of packets, however, presents two problems:
1) with a large number of shaped streams (e.g. more than one data source supplies data packet streams to a network element), a per-stream buffer space may be limited, and
2) queuing increases end-to-end delay between a sending communication side and a receiving communication side. This in turn has adverse effects on quality in the case of VoIP and videoconferencing as experienced by an end user.
Due to this, traffic shapers usually have a limited perstream buffer space combined with burst allowance. Burst allowance is defined as an allowed variation of bandwidth from the nominal bit rate of the stream. Another name for burst allowance is leaky bucket regulator. More precisely, the larger the burst allowance, the larger clusters of packets can pass through the shaper without queuing.
However, dimensioning of the burst allowance for a traffic shaping device is rather difficult, since too large a value for the burst allowance defeats the original purpose of traffic shaping, whereas too small a value for the burst allowance leads to excessive packet losses.
In the latter connection, it has to be noted that a degree of allowable packet loss is determined for a respective application the data of which are transmitted, e.g. for a speech application (voice transmission) by the audio coding used in the media streams. Modern speech codecs can usually tolerate a small amount (less than 1%) of lost frames without clear audible degradation in speech quality. However, several consecutive frame losses will inevitably lead to severe quality problems. In many cases a modern speech codec can tolerate 2–3 adjacent missing frames without excessive degradation in subjective speech quality, provided that packet losses are separated by a large enough number of packets that have arrived at the receiver. (Note that one or more (audio) frames compose one data packet (IP data packet), while, however, one frame includes many data samples.) In listening tests, it has been observed that the mean opinion score (MOS) is roughly proportional to the average number of arrived packets between two packet losses.